Re: [Asterisk-Users] IAX phone not hear the other phone ring when calling
option r. 'nuff said. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+dial On Sat, 23 Jul 2005 14:48:10 -0500, Maps [EMAIL PROTECTED] said: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how does pattern routes works
Pattern-matching extensions must be prefaced with an underscore thus: _1NXXNXX Enjoy! On Fri, 1 Jul 2005 10:32:46 -0700 (PDT), wassim darwish [EMAIL PROTECTED] said: i tried to write to usa destination 1* it worked well but when i tried to specify the number of digits i wrote 1NXXNXX but it did'nt work.can anybody help me please please. Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Pattern-matching extensions must be prefaced with an underscore thus: _1NXXNXX Enjoy! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pattern matching based on callerid, not working
I'm not sure it's the source of your problem, but I'm sure it could wind
up being the source of others:
I think that should be:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,1,Set(CALLERID(number)=281444) ; these should
exten = _9./3005,1,Set(CALLERID(number)=281445) ; all be priority
exten = _9./3006,1,Set(CALLERID(number)=281446) ; 1, not n
exten = _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
On Fri, 01 Jul 2005 13:15:08 -0500, Matthew Boehm
[EMAIL PROTECTED] said:
according to the wiki, I should be able to do this:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,n,Set(CALLERID(number)=281444)
exten = _9./3005,n,Set(CALLERID(number)=281445)
exten = _9./3006,n,Set(CALLERID(number)=281446)
exten = _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
and have the correct calleridnum's set for each extension based on their
current calleridnum.
Basically, priority 1 will execute only if callerid is currently 3003.
pri2 will only execute if callerid is 3004, etc..
however, attempts to do this all fail with auto-fallthru BUSY.
Im using most recent CVS-HEAD.
Any ideas?
-Matthew
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I'm not sure it's the source of your problem, but I'm sure it could wind
up being the source of others:
I think that should be:
exten = _9./3003,1,Set(CALLERID(number)=281443)
exten = _9./3004,1,Set(CALLERID(number)=281444) ; these should
exten = _9./3005,1,Set(CALLERID(number)=281445) ; all be priority
exten = _9./3006,1,Set(CALLERID(number)=281446) ; 1, not n
exten = _9.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,wt)
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Re: [Asterisk-Users] Giving user progress in an voice menu system
(here at the top because I can never remember which placement of the reply is preferred) Your 'macro to dial the phones' presumably has a Dial statement...add the option 'r' or 'm' (qq.v. in show application dial) thereto. On Thu, 12 May 2005 12:43:47 -0700, Sean Kennedy [EMAIL PROTECTED] said: Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is apprecaited. Sean [800-in] exten = s,1,Answer exten = s,2,Background(billing-welcome) exten = s,3,ResponseTimeout(5) exten = s,4,Background(billing-menu) exten = t,1,Goto(s,3) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,2) exten = 101,1,Ringing exten = 101,2,Wait(1) exten = 101,3,Macro(ext,101) exten = 113,1,Ringing exten = 113,2,Wait(1) exten = 113,3,Macro(ext,113) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users (here at the bottom because I can never remember which placement of the reply is preferred) Your 'macro to dial the phones' presumably has a Dial statement...add the option 'r' or 'm' (qq.v. in show application dial) thereto. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What do you name yours
Mine is called 'blacksun', as that's where it's colo'd. (idiocy in a naming convention, I know.) On Wed, 11 May 2005 19:55:36 -0700 (PDT), Matt Klein [EMAIL PROTECTED] said: Mine is named spike... On Thu, 12 May 2005, Paul Hales wrote: We bought one of those books on the worst cars ever made...every page has great names... PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Thursday, 12 May 2005 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Andrew Latham Subject: Re: [Asterisk-Users] What do you name yours On Wed, 2005-05-11 at 10:09 -0500, Andrew Latham wrote: Naming Conventions for Asterisk Hostnames, . For an internal historical reason all ours come from the legends of Robin Hood. I used to work with a bunch of Lord of the Rings readers and all the machine names came from there. It always makes a good light discussion point. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call initiation
Others correct me if I'm wrong, but I believe that:
Presumably, you have lines like:
exten = _XXX,1,Dial(SIP/${EXTEN})
exten = _9NXX,1,Dial(ZAP/1/${EXTEN:1})
exten = _91NXXNXX,1,Dial(Zap/1/${EXTEN:1})
if none of your internal extensions start with a nine (ie if they are in
the range 000-899 or smaller), try changing the first to:
exten = _[0-8]XX,1,Dial(SIP/${EXTEN})
or even narrower as your needs require
This way, * doesn't have to wait for you to stop dialing to distinguish
between outgoing 918185551212 and extension 918.
Hope this helps!
-Chris
On Fri, 23 Apr 2004 14:39:28 -0500, Roger [EMAIL PROTECTED] said:
Users withing the office can dial a 3 digit extension and that will ring
a phone. The problem I'm running into is you have to press xxx then
press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an
internal extension and automatically dial it the user has to initiate
that call. Asterisk automatically initiates calls w/ 9+7 digits and LD
calls, 9+1+areacode+number.
How would you tell the PBX try an extension once and 3 digits have been
pressed. The exception being 9 as that gives a outside line.
--
Rock River Internet Roger Grunkemeyer
202 W. State St, 8th Floor[EMAIL PROTECTED]
Rockford, IL 61101 815-968-9888 x101
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RE: [Asterisk-Users] VoicePulse Connect Problems
Mine, too, are fixed...I was in much the same boat as the original poster...an old DID in 212 worked with DTMF, two much newer ones in 213 and 818 (new markets, apparently) didn't until this morning. On Tue, 13 Apr 2004 16:02:37 -0400, Robert Jackson [EMAIL PROTECTED] said: Very cool. I am just glad they got it fixed. -Original Message- From: Isaac McDonald [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 3:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoicePulse Connect Problems It works now! I did nothing on my end either. VP must monitor this list. Isaac Robert Jackson wrote: Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls, through VoicePulse of course. It was working properly yesterday, and without changing anything it stopped working. Thanks in advance, Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Live real extensions.conf samples?
I consider good examples to be those of John Todd and Zac Sprackett, viz: http://www.loligo.com/asterisk/current/extensions.conf http://sprackett.com/asterisk/conf/extensions.conf If you lop the filename off each of those, you also get a directory of *all* their .conf files, also good reading. N.B.: In their respective sip.conf's and iax.conf's, while both of them change usernames and passwords to protect the innocent, IMHO, Todd does it in a way which leaves it clearer how to use those files. Good examples especially for the various commercial gateways out there. Hope this helps! -Chris - Original message - From: Ken Godee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sun, 02 Nov 2003 15:35:28 -0700 Subject: [Asterisk-Users] Live real extensions.conf samples? It would be nice to see a real extensions.conf from a live business operation, every extensions.conf I've seen posted or been able to dig up so far would fail bad in a live business operation. I just have the beginings of mine and would like to make sure I don't miss anything. Most extensions.conf files I've seen wouldn't even let you dial 911 in thier dialplan. That's just something you don't want to forget! Not to mention that a business type extensions.conf needs to have several class of restrictions for different departments/people, most just have everything available to everyone, this is just not so in the real world. Not it mine anyway. If someone doesn't want to post you can alway email me direct. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP client
It may not be *exactly* what you're looking for, but try: http://fwd.pulver.com/callme.php?userid=411 In examining the source, it seems you can put any SIP address, not just FWD ones, though there doesn't seem to be any overt SIP registration going on. - Original message - From: Rattana BIV [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 29 Oct 2003 09:58:28 +0100 Subject: [Asterisk-Users] SIP client hi everybody, Is there SIP client which work with Asterisk and can be embedded in a HTML page ? Thanks Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicepulse and IAX
The instructions they sent you (and me) are slightly faulty. in iax.conf: context=VPWS ;I'm not sure what VPWS stands for. would better be context=from-voicepulse ;(for example...this is what I use) then in extensions.conf: (this is the simplest example) ;snip [from-voicepulse] exten=3017275115,1,Dial(SIP/1234) ;obviously replacing 3017275115 with your voicepulse # (if it isn't already) ; and SIP/1234 with your registered SIP client or Zap channel or whatever. ; then this phone will ring when someone calls your number. You could, of course, leave the context as VPWS in iax.conf, just make sure there is a corresponding context in your extensions.conf. An incoming call that no context/extension covers is the source of the NOTICE you mentioned. Hope this helps! Chris - Original message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 29 Oct 2003 17:07:42 + Subject: [Asterisk-Users] Voicepulse and IAX I am trying to set up IAX with Voicepulse. When I turn on debugging I get the following message when I call my PSTN number: NOTICE[1142106560]: File chan_iax2.c, Line 4321 (socket_read): Rejected connect attempt from 66.234.228.132, request '[EMAIL PROTECTED]' does not exist Any help would be GREATLY appreciated. Thanks, Isaac [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
Well, my hosts hack-on-hack didn't work...internal clients could register with * using the hosts-hacked FQDN, and * could register with (for example) FWD and iconnecthere, but on calls in either direction, I only got a few seconds of audio, then silence (though debugging showed what looked like a continued normal SIP/RTP conversation), then it threw a 484 Address Incomplete status and disconnected. Would it be inappropriate to sponsor ($) a dev contest for the real (universal/not hardcoded) 'hack'? :) - Original message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 29 Oct 2003 09:13:31 -0800 (PST) Subject: Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients --- Peter Zeltins [EMAIL PROTECTED] wrote: That's for pointing out Walter Snel hack. Adding his two additional features would not be hard. http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html Any idea when these hacks will appear in CVS? We should all hope never. That's why you call it a hack because it works for only one very specific case and would break SIP under Astrisk for most people. It even breaks calls between Asterisk and local SIP phones. Now the trick is to write some code that desides if the trick is to be used or not for each call by comparing the IP address of Asterisk and the called SIP phone. You migh want to experiment with it and report results. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Exclusive Video Premiere - Britney Spears http://launch.yahoo.com/promos/britneyspears/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients
Hello everyone and welcome to my first post to the list! After studying for a couple of weeks, I finally built * for the first time last night, and of course had the same SIP-behind-NAT woes that plague all of us who use NATted connections. It was therefore with no small joy that I read the fix for that that Walter Snel proposed (q.v.: http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html). Since I currently have no zaptel hardware (though intend to get some within the week) and thus use soft (SIP) clients on the same internal network, the caveat that it would break internal SIP clients was, for me, a reason to not yet implement his fix. I was examining chan_sip.c, trying to think of a way to implement his 'Naturally it would be much better to make this behavior:', and while bemoaning my pathetic C skills, thought of another solution: -CUT HERE TO GET RIGHT TO THE POINT :P- 1) Somehow (I use dynamic DNS) get a FQDN to point to the IP of the outside of your NAT box; 2) Implement W Snel's hardcoding as in the above URL, using your chosen FQDN where, in his example, he has 213.84.4.39; 3) On any internal machine with a SIP client, add an entry to the hosts file that points your chosen FQDN to the * server's IP on the *internal* network. What do you all think? I won't have an opportunity to try to implement this until later this evening, but at that time will post a follow-up to let you know how it went. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
